similar to: Asterisk SIP authentication against [section] instead of username

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

I have two Asterisks connected using SIP. One is acting as a SIP "server", the other as a SIP "client". This almost works; but calls from 50607795 are rejected with this error: check_auth: username mismatch, have <50607796>, digest has <50607795> On the "client" I have these accounts configured in sip.conf: register => 50607795:test at

1.6.2.13, sip.conf: [155] type=friend context=longdistance callerid="Admin" <155> secret=test host=dynamic dtmfmode=rfc2833 allow=all defaultuser=155-trust ............ On aastra: Basic SIP Authentication Settings Screen Name Phone Number 155 Caller ID 155 Authentication Name 155-trust Password test But: WARNING[1737]:

Hello, everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get the following : [Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username mismatch, have <329909006666>, digest has <3291119600> [Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite: Failed to authenticate device "0473990000" <sip:0473990000 at

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

In SIP setting on the e71 I set the public user name as 1995 at 10.10.11.180. There is a sip.conf context [1995] On the asterisk CLI I get: Registration from '<sip:%201995 at 10.10.11.180:5060>' failed for '10.10.11.98' - No matching peer found So I changed the sip.conf context to [%201995] Then: [2009-11-19 20:44:28] WARNING[14371]: chan_sip.c:11797 check_auth:

Hi all, I wrote a small script (developed and testet on Solaris 8), which automates the generation and installation of the steps needed to put keys in place. I you are interested to take it, feel free to do it. -- *** Freundliche Gruesse **** Best regards *** Anton Burkhalter Dipl. El. Ing. HTL Mobile:+41(0)78 844-0290 mailto:anton.burkhalter at gmx.net

Le 17/04/2018 ? 14:18, kfx a ?crit?: > dovecot 2.2.34 > solr 7.2 > > I only see new messages after typing on the server "doveadm fts rescan > -u username" though I've followed the wiki and added "fts_autoindex = > yes" in 90-plugin.conf . Subsequent search for the same pattern always > gives the same result, ignoring new emails with that particular

Hi, I'm trying to get the match_auth_username=yes sip configuration working. It's mentioned as an experimental new feature of 1.6.2.x. (I'm using 1.6.2.8) The sip.conf example states: ; if available, match user entry using the ; 'username' field from the authentication line ; instead of the From: field. But still I've been unable to authenticate using username

Hi, Andrew. You are trying to solve two tasks: definition through what line the call came and a beautiful display of this information. 1. definition through what line the call came. If the username and password for inbound and outbound registration the same, then try the following: a) delete "register" lines. b) add option "callbackextension=Company1" to Company1 friend

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but it does work. For prosperity, the SIP service is through Internode. Here is my "extensions.conf" file: exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one exten => s,2,Set(pseudodid=${SIP_HEADER(To)}) exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)}) exten =>

I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454

Hello, I'm trying to receive a fax to my inbound number from voipbuster. Asterisk receives the call and starts the rxfax application successful, but then nothing happens. The calling party is still hearing a ringing tone, or sometimes nothing. Voicecalls are working correct and without problems. For testing I've add a local number (300) to the dialplan. When I call this number

I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While everything seems to be working fine I keep getting this error on my log files: [2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.2.23', "port" = '5060', "regseconds" = '1392692118',

I created an account at IAXTEL.com to route 1-700-XXX-XXXX calls through. IAXTEL.com gave me a number (example) of 700-555-6226. I have made the following changes to my: /etc/asterisk/extensions.conf: [iaxtel700] exten => _81700XXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1}) exten => _81800NXXXXXX,1,Dial(IAX2/myusername:mypassword@iaxtel.com/${EXTEN:1})

Hi Guys, I have Spikko setup as provider of DID and outbound routes and I can make calls out but no inbound calls via DID can be made. I did a sip debug which is reported below. I never receive the call though, I have a catch all in my inbound routes and it doesn't hit my context at all or not sip invite comes in: FreePBX: Trunk Name: *Spikko* Peer Detail *username=MyUsername*

Hi I set up a samba 4.1.4 server on the latest FreeBSD RELEASE 10. Unfortunately it doesn't seem to consider the option force group. After hours ofresearch I couldn't figure out what I'm still missing. unix extensions is set to no. Setting the debug level up to 10 also didn't help ;( Is this a bug or is there simply a mistake in my setup? When *valid users = @Groupname* is

Running wine 0.9.28, Ubuntu Edgy. nvidia 6800GT, dual-LCD w/ xinerama, nv-glx driver. Running winecfg always returns: wine: creating configuration directory '/home/myusername/.wine'... X Error of failed request: BadWindow (invalid Window parameter) Major opcode of failed request: 144 (NV-GLX) Minor opcode of failed request: 4 () Resource id in failed request: 0x24a Serial

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29]